5
means the change in sampling rate employed in today’s modern D/A converter
chips themselves.
But let’s start at the beginning. What is the sampling frequency? For any
digital storage or transmission it is necessary to have time discrete samples of
the signal which has to be processed. I.e. the analog signal has to be sampled at
discrete time intervals and later converted to digital numbers (also see section
Jitter Suppression and Clocking
, p. 3). This sampling and conversion process
happens in the so-called analog to digital converter (A/D). The inverse in the
digital to analog converter (D/A).
A physical law states that in order to represent any given analog signal in the
digital domain, one has to sample that signal with at least twice the frequency
of the highest frequency contained in the analog signal. If this law is violated so
called aliasing components are generated which are perceived as a very nasty
kind of distortion. So if one defines the audio band of interest to lie between
20 Hz and 20 kHz, then the minimum sampling frequency for such signals must
be 40 kHz.
For practical reasons explained below, the sampling frequency of 44.1 kHz
was chosen for the CD. A sampling frequency of 44.1 kHz allows to represent
signals up to 22.05 kHz. The designer of the system has to take care that
any frequencies above 22.05 kHz are sufficiently suppressed before sampling at
44.1 kHz. This suppression is done with the help of a low pass filter, which
cuts off the frequencies above 22.05 kHz. In practice such a filter has a limited
steepness, i.e. if it suppresses frequencies above 22.05 kHz it also suppresses
frequencies between 20 kHz and 22.05 kHz to some extent. So in order to have a
filter which sufficiently suppresses frequencies above 22.05 kHz, one has to allow
it to have a so-called transition band between 20 kHz and 22.05 kHz where it
gradually builds up its suppression.
Note that so far we have talked about the so-called anti-aliasing filter, which
filters the audio signal ahead of the A/D conversion process. For the D/A
conversion, which is of more interest to the High-End Hi-Fi enthusiast, essentially
the same filter is required. This is because after the D/A conversion we have a
time discrete analog signal, i.e. a signal that looks like steps, having the rate of
the sampling frequency.
Such a signal contains not only the original audio signal between 20 Hz and
20 kHz but also replicas of the same signal symmetrical around multiples of the
sampling frequency. This may sound complicated, but the essence is that there
are now signals above 22.05 kHz. These signals come from the sampling process.
There are now frequencies above 22.05 kHz which have to be suppressed, so that
they do not cause any intermodulation distortion in the amplifiers and speakers,
do not burn tweeters or do not make the dog go mad.
Again, a low pass filter, which is called a
”
reconstruction filter“, is here to
suppress those frequencies. The same applies to the reconstruction filter as to the
anti-aliasing filter: pass-band up to 20 kHz, transition-band between 20 kHz and
22.05 kHz, stop-band above 22.05 kHz. You may think that such a filter is rather
Summary of Contents for INT204
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